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| -rw-r--r-- | README.md | 25 |
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@@ -157,6 +157,31 @@ name for the encoder: The samplerate of the JACK server should be 32kHz or 48kHz. +Scenario *LiveWire* or *AES67* +------------------------------ + +When audio data is available on the network as a multicast stream, it can be encoded using the following pipeline: + + rtpdump -F payload 239.192.1.1/5004 | \ + sox -t raw -e signed-integer -r 48000 -c 2 -b 24 -B /dev/stdin -t raw --no-dither -r 48000 -c 2 -b 16 -L /dev/stdout gain 4 | \ + odr-audioenc -f raw -b $BITRATE -i /dev/stdin -o $DST + +It is also possible to use the libvlc input, where you need to create an SDP file with the following contents: + + v=0 + o=Node 1 1 IN IP4 172.16.235.155 + s=TestSine + t=0 0 + a=type:multicast + c=IN IP4 239.192.0.1 + m=audio 5004 RTP/AVP 97 + a=rtpmap:97 L24/48000/2 + +Replace the IP address in the `o=` field by the one corresponding to your +source node IP address, and the IP in `c=` by the multicast IP of your stream. +Then use this SDP file as input for the VLC input. + + Scenario *local file through snd-aloop* --------------------------------------- Play some local audio source from a file, with ZMQ output for ODR-DabMux. The problem with |